쥐스킨트의 작은 서재

Stream Control Transmission Protocol

In computer networking, the Stream Control Transmission Protocol (SCTP) is a transport-layer protocol, serving in a similar role to the popular protocols TCP and UDP.

SCTP provides some of the same service features of both: it is message-oriented like UDP and ensures reliable, in-sequence transport of messages with congestion control like TCP; it differs from these in providing multi-homing and redundant paths to increase resilience and reliability.

In the absence of native SCTP support in operating systems it is possible to tunnel SCTP over UDP, as well as mapping TCP API calls to SCTP ones. The reference implementation was released as part of FreeBSD version 7. It has subsequently been widely ported.

Real-time Transport Protocol

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction witha signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network.

Overview

RTP is designed for end-to-end, real-time, transfer of streaming media. The protocol provides facilities for jitter compensation and detection of out of sequence arrival in data, which are common during transmission on an IP network. RTP allows data transfer to multiple destinations through IP multicast. RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.

Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet lossto achieve this goal. For example, loss of a packet in audio application may result in loss of a fraction of a second of audio data, which can be made unnoticeable with suitable error concealment algorithms. The Transmission Control Protocol (TCP), although standardized for RTP use, is not normally used in RTP applications because TCP favors reliability over timeliness. Instead themajority of the RTP implementations are built on the User Datagram Protocol(UDP). Other transport protocols specifically designed for multimedia sessions are SCTP and DCCP, although, as of 2010, they are not in widespread use.

Protocol components

The RTP specification describes two sub-protocols, RTP and RTCP.

The data transfer protocol, RTP, facilitates the transfer of real-time data. Information provided by this protocol include timestamps, sequence numbers and the payload format which indicates the encoded format of the data.

The control protocol RTCP is used tospecify quality of service (QoS) feedback and synchronization between the media streams. The bandwidth of RTCP traffic compared to RTP is small, typically around 5%.

User Datagram Protocol

The User Datagram Protocol (UDP) is one ofthe core members of the Internet protocol suite. The protocol was designed by David P. Reed in 1980 and formally defined in RFC 768.

UDP uses a simple connectionless transmission model with a minimum of protocol mechanism. It has no handshaking dialogues, and thus exposes the user’s program to any unreliability of the underlying network protocol. There is no guarantee of delivery, ordering, or duplicate protection. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram.

With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths. UDP is suitable for purposes where error checking and correction is either not necessary or is performed in the application, avoiding the overhead of such processing atthe network interface level. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for delayed packets, which may not be an option in a real-time system. If error correction facilities are needed atthe network interface level, an application may use the Transmission Control Protocol (TCP) or Streaming Control Transmission Protocol (SCTP) which are designed for this purpose.

Packet structure

UDP is a minimal message-oriented Transport Layer protocol that is documented in IETF RFC 768.

UDP provides no guarantees to the upper layer protocol for message delivery and the UDP layer retains no state of UDP messages once sent. For this reason, UDP sometimes is referred to as Unreliable Datagram Protocol.

UDP provides application multiplexing (via port numbers) and integrity verification (via checksum) of the header and payload. If transmission reliability is desired, it must be implemented in the user’s application.

The UDP header consists of 4 fields, each of which is 2 bytes (16 bits). The use of the fields “Checksum” and “Source port” is optional in IPv4. In IPv6 only the source port is optional.

-      Source port number: This field identifies the sender’s port when meaningful and should be assumed to be the port to reply to if needed. If not used, then it should be zero. If the source host is the client, the port number is likely to be an ephemeral port number.

-      Destination port number: This field identifies the receiver’s port and is required. Similar to source port number, if the client is the destination host then the port number will likely be an ephemeral port number and if the destination host is the server then the port number will likely be a well-known port number.

-      Length: A field that specifies the length in bytes of the UDP header and UDP data. The minimum length is 8 bytes because that is the length of the header. The field size sets a theoretical limit of 65,535 bytes (8 byte header + 65,527 bytes of data) for a UDP datagram. The practical limit for the data length which is imposed by the underlying IPv4 protocol is 65,507 bytes (65,535 – 8 byte UDP header – 20 byte IP header).

-      Checksum: The checksum field may be used for error-checking of the header and data. This field is optional in IPv4, and mandatory in IPv6. The field carries all-zeros if unused.

Reliability and congestion control solutions

Lacking reliability, UDP applications must generally be willing to accept some loss, errors or duplication. Some applications, such as TFTP, may add rudimentary reliability mechanisms into the application layer as needed.

Most often, UDP applications do not employer liability mechanisms and may even be hindered by them. Streaming media, real-time multiplayer games and voice over IP (VoIP) are examples of applications that often use UDP. In the separticular applications, loss of packets is not usually a fatal problem. If anapplication requires a high degree of reliability, a protocol such as the Transmission Control Protocol may be used instead.

From: https://en.wikipedia.org/wiki/User_Datagram_Protocol